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Versions: (draft-portman-siprec-protocol) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 RFC 7866

SIPREC                                                        L. Portman
Internet-Draft                                              NICE Systems
Intended status: Standards Track                             H. Lum, Ed.
Expires: March 28, 2016                                          Genesys
                                                                C. Eckel
                                                                   Cisco
                                                             A. Johnston
                                                                   Avaya
                                                               A. Hutton
                                                                   Unify
                                                      September 25, 2015


                       Session Recording Protocol
                     draft-ietf-siprec-protocol-18

Abstract

   This document specifies the use of the Session Initiation Protocol
   (SIP), the Session Description Protocol (SDP), and the Real Time
   Protocol (RTP) for delivering real-time media and metadata from a
   Communication Session (CS) to a recording device.  The Session
   Recording Protocol specifies the use of SIP, SDP, and RTP to
   establish a Recording Session (RS) between the Session Recording
   Client (SRC), which is on the path of the CS, and a Session Recording
   Server (SRS) at the recording device.  This document considers only
   active recording, where the SRC purposefully streams media to an SRS
   and all participating user agents are notified of the recording.
   Passive recording, where a recording device detects media directly
   from the network (e.g., using port-mirroring techniques), is outside
   the scope of this document.  In addition, lawful intercept is outside
   the scope of this document.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."




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   This Internet-Draft will expire on March 28, 2016.

Copyright Notice

   Copyright (c) 2015 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  Definitions . . . . . . . . . . . . . . . . . . . . . . . . .   4
   4.  Scope . . . . . . . . . . . . . . . . . . . . . . . . . . . .   4
   5.  Overview of operations  . . . . . . . . . . . . . . . . . . .   5
     5.1.  Delivering recorded media . . . . . . . . . . . . . . . .   5
     5.2.  Delivering recording metadata . . . . . . . . . . . . . .   8
     5.3.  Receiving recording indications and providing recording
           preferences . . . . . . . . . . . . . . . . . . . . . . .   9
   6.  SIP Handling  . . . . . . . . . . . . . . . . . . . . . . . .  11
     6.1.  Procedures at the SRC . . . . . . . . . . . . . . . . . .  11
       6.1.1.  Initiating a Recording Session  . . . . . . . . . . .  11
       6.1.2.  SIP extensions for recording indication and
               preference  . . . . . . . . . . . . . . . . . . . . .  11
     6.2.  Procedures at the SRS . . . . . . . . . . . . . . . . . .  12
     6.3.  Procedures for Recording-aware User Agents  . . . . . . .  12
   7.  SDP Handling  . . . . . . . . . . . . . . . . . . . . . . . .  13
     7.1.  Procedures at the SRC . . . . . . . . . . . . . . . . . .  13
       7.1.1.  SDP handling in RS  . . . . . . . . . . . . . . . . .  13
         7.1.1.1.  Handling media stream updates . . . . . . . . . .  14
       7.1.2.  Recording indication in CS  . . . . . . . . . . . . .  15
       7.1.3.  Recording preference in CS  . . . . . . . . . . . . .  16
     7.2.  Procedures at the SRS . . . . . . . . . . . . . . . . . .  16
     7.3.  Procedures for Recording-aware User Agents  . . . . . . .  18
       7.3.1.  Recording indication  . . . . . . . . . . . . . . . .  18
       7.3.2.  Recording preference  . . . . . . . . . . . . . . . .  19
   8.  RTP Handling  . . . . . . . . . . . . . . . . . . . . . . . .  20
     8.1.  RTP Mechanisms  . . . . . . . . . . . . . . . . . . . . .  20
       8.1.1.  RTCP  . . . . . . . . . . . . . . . . . . . . . . . .  20



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       8.1.2.  RTP Profile . . . . . . . . . . . . . . . . . . . . .  21
       8.1.3.  SSRC  . . . . . . . . . . . . . . . . . . . . . . . .  21
       8.1.4.  CSRC  . . . . . . . . . . . . . . . . . . . . . . . .  22
       8.1.5.  SDES  . . . . . . . . . . . . . . . . . . . . . . . .  22
         8.1.5.1.  CNAME . . . . . . . . . . . . . . . . . . . . . .  22
       8.1.6.  Keepalive . . . . . . . . . . . . . . . . . . . . . .  22
       8.1.7.  RTCP Feedback Messages  . . . . . . . . . . . . . . .  23
         8.1.7.1.  Full Intra Request  . . . . . . . . . . . . . . .  23
         8.1.7.2.  Picture Loss Indicator  . . . . . . . . . . . . .  23
         8.1.7.3.  Temporary Maximum Media Stream Bit Rate Request .  24
       8.1.8.  Symmetric RTP/RTCP for Sending and Receiving  . . . .  24
     8.2.  Roles . . . . . . . . . . . . . . . . . . . . . . . . . .  25
       8.2.1.  SRC acting as an RTP Translator . . . . . . . . . . .  26
         8.2.1.1.  Forwarding Translator . . . . . . . . . . . . . .  26
         8.2.1.2.  Transcoding Translator  . . . . . . . . . . . . .  26
       8.2.2.  SRC acting as an RTP Mixer  . . . . . . . . . . . . .  27
       8.2.3.  SRC acting as an RTP Endpoint . . . . . . . . . . . .  28
     8.3.  RTP Session Usage by SRC  . . . . . . . . . . . . . . . .  28
       8.3.1.  SRC Using Multiple m-lines  . . . . . . . . . . . . .  28
       8.3.2.  SRC Using Mixing  . . . . . . . . . . . . . . . . . .  29
     8.4.  RTP Session Usage by SRS  . . . . . . . . . . . . . . . .  30
   9.  Metadata  . . . . . . . . . . . . . . . . . . . . . . . . . .  31
     9.1.  Procedures at the SRC . . . . . . . . . . . . . . . . . .  31
     9.2.  Procedures at the SRS . . . . . . . . . . . . . . . . . .  33
   10. Persistent Recording  . . . . . . . . . . . . . . . . . . . .  35
   11. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  35
     11.1.  Registration of Option Tags  . . . . . . . . . . . . . .  35
       11.1.1.  siprec Option Tag  . . . . . . . . . . . . . . . . .  35
       11.1.2.  record-aware Option Tag  . . . . . . . . . . . . . .  36
     11.2.  Registration of media feature tags . . . . . . . . . . .  36
       11.2.1.  src feature tag  . . . . . . . . . . . . . . . . . .  36
       11.2.2.  srs feature tag  . . . . . . . . . . . . . . . . . .  36
     11.3.  New Content-Disposition Parameter Registrations  . . . .  37
     11.4.  Media Type Registration  . . . . . . . . . . . . . . . .  37
     11.5.  SDP Attributes . . . . . . . . . . . . . . . . . . . . .  37
       11.5.1.  'record' SDP Attribute . . . . . . . . . . . . . . .  37
       11.5.2.  'recordpref' SDP Attribute . . . . . . . . . . . . .  38
   12. Security Considerations . . . . . . . . . . . . . . . . . . .  38
     12.1.  Authentication and Authorization . . . . . . . . . . . .  39
     12.2.  RTP handling . . . . . . . . . . . . . . . . . . . . . .  39
     12.3.  Metadata . . . . . . . . . . . . . . . . . . . . . . . .  40
     12.4.  Storage and playback . . . . . . . . . . . . . . . . . .  40
   13. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  41
   14. References  . . . . . . . . . . . . . . . . . . . . . . . . .  41
     14.1.  Normative References . . . . . . . . . . . . . . . . . .  41
     14.2.  Informative References . . . . . . . . . . . . . . . . .  42
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  44




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1.  Introduction

   This document specifies the mechanism to record a Communication
   Session (CS) by delivering real-time media and metadata from the CS
   to a recording device.  In accordance with the architecture
   [RFC7245], the Session Recording Protocol specifies the use of SIP,
   SDP, and RTP to establish a Recording Session (RS) between the
   Session Recording Client (SRC), which is on the path of the CS, and a
   Session Recording Server (SRS) at the recording device.  SIP is also
   used to deliver metadata to the recording device, as specified in
   [I-D.ietf-siprec-metadata].  Metadata is information that describes
   recorded media and the CS to which they relate.  The Session
   Recording Protocol intends to satisfy the SIP-based Media Recording
   requirements listed in [RFC6341].  In addition to the Session
   Recording Protocol, this document specifies extensions for user
   agents that are participants in a CS to receive recording indications
   and to provide preferences for recording.

   This document considers only active recording, where the SRC
   purposefully streams media to an SRS and all participating user
   agents are notified of the recording.  Passive recording, where a
   recording device detects media directly from the network (e.g., using
   port-mirroring techniques), is outside the scope of this document.
   In addition, lawful intercept is outside the scope of this document,
   in accordance with [RFC2804].

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

3.  Definitions

   This document refers to the core definitions provided in the
   architecture document [RFC7245].

   The RTP Handling section uses the definitions provided in "RTP: A
   Transport Protocol for Real-Time Application" [RFC3550].

4.  Scope

   The scope of the Session Recording Protocol includes the
   establishment of the recording sessions and the reporting of the
   metadata.  The scope also includes extensions supported by User
   Agents participating in the CS such as indication of recording.  The
   user agents need not be recording-aware in order to participate in a
   CS being recorded.



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   The following items, which are not an exhaustive list, do not
   represent the protocol itself and are considered out of the scope of
   the Session Recording Protocol:

   o  Delivering recorded media in real-time as the CS media

   o  Specifications of criteria to select a specific CS to be recorded
      or triggers to record a certain CS in the future

   o  Recording policies that determine whether the CS should be
      recorded and whether parts of the CS are to be recorded

   o  Retention policies that determine how long a recording is stored

   o  Searching and accessing the recorded media and metadata

   o  Policies governing how CS users are made aware of recording

   o  Delivering additional recording session metadata through a non-SIP
      mechanism

5.  Overview of operations

   This section is informative and provides a description of recording
   operations.

   Section 6 describes the SIP communication in a recording session
   between an SRC and an SRS, and the procedures for recording-aware
   user agents participating in a CS.  Section 7 describes the SDP in a
   recording session, and the procedures for recording indications and
   recording preferences.  Section 8 describes the RTP handling in a
   recording session.  Section 9 describes the mechanism to deliver
   recording metadata from the SRC to the SRS.

   As mentioned in the architecture document [RFC7245], there are a
   number of types of call flows based on the location of the Session
   Recording Client.  The following sample call flows provide a quick
   overview of the operations between the SRC and the SRS.

5.1.  Delivering recorded media

   When a SIP Back-to-Back User Agent (B2BUA) with SRC functionality
   routes a call from UA(A) to UA(B), the SRC has access to the media
   path between the user agents.  When the SRC is aware that it should
   be recording the conversation, the SRC can cause the B2BUA to relay
   the media between UA(A) and UA(B).  The SRC then establishes the
   Recording Session with the SRS and sends replicated media towards the
   SRS.



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   An endpoint may also have SRC functionality, where the endpoint
   itself establishes the Recording Session to the SRS.  Since the
   endpoint has access to the media in the Communication Session, the
   endpoint can send replicated media towards the SRS.

   The following example call flows shows an SRC establishing a
   recording session towards an SRS.  The first call flow illustrates
   UA(A) acting as the SRC.  The second illustrates a B2BUA acting as
   the SRC.  Note that the SRC can choose when to establish the
   Recording Session independent of the Communication Session, even
   though the following call flows suggest that the SRC is establishing
   the Recording Session (message #5) after the Communication Session is
   established.


            UA A/SRC               UA B                    SRS
             |(1)CS INVITE           |                      |
             |---------------------->|                      |
             |           (2) 200 OK  |                      |
             |<----------------------|                      |
             |                       |                      |
             |(3)RS INVITE with SDP  |                      |
             |--------------------------------------------->|
             |                       |  (4) 200 OK with SDP |
             |<---------------------------------------------|
             |(5)CS RTP              |                      |
             |======================>|                      |
             |<======================|                      |
             |(6)RS RTP              |                      |
             |=============================================>|
             |=============================================>|
             |                       |                      |
             |(7)CS BYE              |                      |
             |---------------------->|                      |
             |(8)RS BYE              |                      |
             |--------------------------------------------->|
             |                       |                      |


            Figure 1: Basic recording call flow with UA as SRC











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     UA A           SRC                    UA B                    SRS
      |(1)CS INVITE  |                       |                      |
      |------------->|                       |                      |
      |              |(2)CS INVITE           |                      |
      |              |---------------------->|                      |
      |              |           (3) 200 OK  |                      |
      |              |<----------------------|                      |
      |   (4) 200 OK |                       |                      |
      |<-------------|                       |                      |
      |              |(5)RS INVITE with SDP  |                      |
      |              |--------------------------------------------->|
      |              |                       |  (6) 200 OK with SDP |
      |              |<---------------------------------------------|
      |(7)CS RTP     |                       |                      |
      |=============>|======================>|                      |
      |<=============|<======================|                      |
      |              |(8)RS RTP              |                      |
      |              |=============================================>|
      |              |=============================================>|
      |(9)CS BYE     |                       |                      |
      |------------->|                       |                      |
      |              |(10)CS BYE             |                      |
      |              |---------------------->|                      |
      |              |(11)RS BYE             |                      |
      |              |--------------------------------------------->|
      |              |                       |                      |


           Figure 2: Basic recording call flow with B2BUA as SRC

   The above call flow can also apply to the case of a centralized
   conference with a mixer.  For clarity, ACKs to INVITEs and 200 OKs to
   BYEs are not shown.  The conference focus can provide the SRC
   functionality since the conference focus has access to all the media
   from each conference participant.  When a recording is requested, the
   SRC delivers the metadata and the media streams to the SRS.  Since
   the conference focus has access to a mixer, the SRC may choose to mix
   the media streams from all participants as a single mixed media
   stream towards the SRS.

   An SRC can use a single recording session to record multiple
   communication sessions.  Every time the SRC wants to record a new
   call, the SRC updates the recording session with a new SDP offer to
   add new recorded streams to the recording session, and
   correspondingly also update the metadata for the new call.






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   An SRS can also establish a recording session to an SRC, although it
   is beyond the scope of this document to define how an SRS would
   specify which calls to record.

5.2.  Delivering recording metadata

   The SRC is responsible for the delivery of metadata to the SRS.  The
   SRC may provide an initial metadata snapshot about recorded media
   streams in the initial INVITE content in the recording session.
   Subsequent metadata updates can be represented as a stream of events
   in UPDATE [RFC3311] or reINVITE requests sent by the SRC.  These
   metadata updates are normally incremental updates to the initial
   metadata snapshot to optimize on the size of updates.  However, the
   SRC may also decide to send a new metadata snapshot any time.

   Metadata is transported in the body of INVITE or UPDATE messages.
   Certain metadata, such as the attributes of the recorded media
   stream, are located in the SDP of the recording session.

   The SRS has the ability to send a request to the SRC to request for a
   new metadata snapshot update from the SRC.  This can happen when the
   SRS fails to understand the current stream of incremental updates for
   whatever reason, for example, when the SRS loses the current state
   due to internal failure.  The SRS may optionally attach a reason
   along with the snapshot request.  This request allows both SRC and
   SRS to synchronize the states with a new metadata snapshot so that
   further metadata incremental updates will be based on the latest
   metadata snapshot.  Similar to the metadata content, the metadata
   snapshot request is transported as content in UPDATE or INVITE sent
   by the SRS in the recording session.





















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          SRC                                                   SRS
           |                                                     |
           |(1) INVITE (metadata snapshot 1)                     |
           |---------------------------------------------------->|
           |                                           (2)200 OK |
           |<----------------------------------------------------|
           |(3) ACK                                              |
           |---------------------------------------------------->|
           |(4) RTP                                              |
           |====================================================>|
           |====================================================>|
           |(5) UPDATE (metadata update 1)                       |
           |---------------------------------------------------->|
           |                                          (6) 200 OK |
           |<----------------------------------------------------|
           |(7) UPDATE (metadata update 2)                       |
           |---------------------------------------------------->|
           |                                          (8) 200 OK |
           |<----------------------------------------------------|
           |              (9) UPDATE (metadata snapshot request) |
           |<----------------------------------------------------|
           |                                        (10) 200 OK  |
           |---------------------------------------------------->|
           |      (11) INVITE (metadata snapshot 2 + SDP offer)  |
           |---------------------------------------------------->|
           |                            (12) 200 OK (SDP answer) |
           |<----------------------------------------------------|
           | (13) UPDATE (metadata update 1 based on snapshot 2) |
           |---------------------------------------------------->|
           |                                         (14) 200 OK |
           |<----------------------------------------------------|


               Figure 3: Delivering metadata via SIP UPDATE

5.3.  Receiving recording indications and providing recording
      preferences

   The SRC is responsible to provide recording indications to the
   participants in the CS.  A recording-aware UA supports receiving
   recording indications via the SDP attribute a=record, and it can
   specify a recording preference in the CS by including the SDP
   attribute a=recordpref.  The recording attribute is a declaration by
   the SRC in the CS to indicate whether recording is taking place.  The
   recording preference attribute is a declaration by the recording-
   aware UA in the CS to indicate its recording preference.  A UA that
   does not want to be recorded may still be notified recording is
   occurring for a number of reasons (e.g., it was not capable of



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   indicating its preference, its preference was ignored, etc.)  If this
   occurs, the UA's only mechanism to avoid being recorded is to
   terminate its participation in the session.

   To illustrate how the attributes are used, if a UA (A) is initiating
   a call to UA (B) and UA (A) is also an SRC that is performing the
   recording, then UA (A) provides the recording indication in the SDP
   offer with a=record:on.  Since UA (A) is the SRC, UA (A) receives the
   recording indication from the SRC directly.  When UA (B) receives the
   SDP offer, UA (B) will see that recording is happening on the other
   endpoint of this session.  Since UA (B) is not an SRC and does not
   provide any recording preference, the SDP answer does not contain
   a=record nor a=recordpref.


        UA A                                                   UA B
        (SRC)                                                   |
          |                                                     |
          |                [SRC recording starts]               |
          |(1) INVITE (SDP offer + a=record:on)                 |
          |---------------------------------------------------->|
          |                             (2) 200 OK (SDP answer) |
          |<----------------------------------------------------|
          |(3) ACK                                              |
          |---------------------------------------------------->|
          |(4) RTP                                              |
          |<===================================================>|
          |                                                     |
          |   [UA B wants to set preference to no recording]    |
          |           (5) INVITE (SDP offer + a=recordpref:off) |
          |<----------------------------------------------------|
          |   [SRC honors the preference and stops recording]   |
          |(6) 200 OK (SDP answer + a=record:off)               |
          |---------------------------------------------------->|
          |                                             (7) ACK |
          |<----------------------------------------------------|


          Figure 4: Recording indication and recording preference

   After the call is established and recording is in progress, UA (B)
   later decides to change the recording preference to no recording and
   sends a reINVITE with the a=recordpref attribute.  It is up to the
   SRC to honor the preference, and in this case SRC decides to stop the
   recording and updates the recording indication in the SDP answer.

   Note that UA (B) could have explicitly indicated a recording
   preference in (2), the 200 OK for the original INVITE.  Indicating a



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   preference of no recording in an initial INVITE or an initial
   response to an INVITE may reduce the chance of a user being recorded
   in the first place.

6.  SIP Handling

6.1.  Procedures at the SRC

6.1.1.  Initiating a Recording Session

   A recording session is a SIP session with specific extensions
   applied, and these extensions are listed in the procedures for SRC
   and SRS below.  When an SRC or an SRS receives a SIP session that is
   not a recording session, it is up to the SRC or the SRS to determine
   what to do with the SIP session.

   The SRC can initiate a recording session by sending a SIP INVITE
   request to the SRS.  The SRC and the SRS are identified in the From
   and To headers, respectively.

   The SRC MUST include the '+sip.src' feature tag in the Contact URI,
   defined in this specification as an extension to [RFC3840], for all
   recording sessions.  An SRS uses the presence of the '+sip.src'
   feature tag in dialog creating and modifying requests and responses
   to confirm that the dialog being created is for the purpose of a
   Recording Session.  In addition, when an SRC sends a REGISTER request
   to a registrar, the SRC MAY include the '+sip.src' feature tag to
   indicate the that it is an SRC.

   Since SIP Caller Preferences extensions are optional to implement for
   routing proxies, there is no guarantee that a recording session will
   be routed to an SRC or SRS.  A new options tag is introduced:
   "siprec".  As per [RFC3261], only an SRC or an SRS can accept this
   option tag in a recording session.  An SRC MUST include the "siprec"
   option tag in the Require header when initiating a Recording Session
   so that UA's which do not support the session recording protocol
   extensions will simply reject the INVITE request with a 420 Bad
   Extension.

   When an SRC receives a new INVITE, the SRC MUST only consider the SIP
   session as a recording session when both the '+sip.srs' feature tag
   and 'siprec' option tag are included in the INVITE request.

6.1.2.  SIP extensions for recording indication and preference

   For the communication session, the SRC MUST provide recording
   indications to all participants in the CS.  A participant UA in a CS
   can indicate that it is recording-aware by providing the "record-



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   aware" option tag, and the SRC MUST provide recording indications in
   the new SDP a=record attribute described in the SDP Handling section.
   In the absence of the "record-aware" option tag, meaning that the
   participant UA is not recording-aware, an SRC MUST provide recording
   indications through other means, such as playing a tone in-band,
   having a signed participant contract in place, etc.

   An SRC in the CS may also indicate itself as a session recording
   client by including the '+sip.src' feature tag.  A recording-aware
   participant can learn that an SRC is in the CS, and can set the
   recording preference for the CS with the new SDP a=recordpref
   attribute described in the SDP Handling section below.

6.2.  Procedures at the SRS

   When an SRS receives a new INVITE, the SRS MUST only consider the SIP
   session as a recording session when both the '+sip.src' feature tag
   and 'siprec' option tag are included in the INVITE request.

   The SRS can initiate a recording session by sending a SIP INVITE
   request to the SRC.  The SRS and the SRC are identified in the From
   and To headers, respectively.

   The SRS MUST include the '+sip.srs' feature tag in the Contact URI,
   as per [RFC3840], for all recording sessions.  An SRC uses the
   presence of this feature tag in dialog creating and modifying
   requests and responses to confirm that the dialog being created is
   for the purpose of a Recording Session (REQ-30).  In addition, when
   an SRS sends a REGISTER request to a registrar, the SRS SHOULD
   include the '+sip.srs' feature tag to indicate that it is an SRS.

   An SRS MUST include the "siprec" option tag in the Require header as
   per [RFC3261] when initiating a Recording Session so that UA's which
   do not support the session recording protocol extensions will simply
   reject the INVITE request with a 420 Bad Extension.

6.3.  Procedures for Recording-aware User Agents

   A recording-aware user agent is a participant in the CS that supports
   the SIP and SDP extensions for receiving recording indications and
   for requesting recording preferences for the call.  A recording-aware
   UA MUST indicate that it can accept reporting of recording indication
   provided by the SRC with a new option tag "record-aware" when
   initiating or establishing a CS, meaning including the "record-aware"
   tag in the Supported header in the initial INVITE request or
   response.





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   A recording-aware UA MUST provide a recording indication to the end
   user through an appropriate user interface, indicating whether
   recording is on, off, or paused for each medium.  Appropriate user
   interfaces may include real-time notification or previously
   established agreements that use of the device is subject to
   recording.  Some user agents that are automatons (e.g., IVR, media
   server, PSTN gateway) may not have a user interface to render
   recording indication.  When such a user agent indicates recording
   awareness, the UA SHOULD render recording indication through other
   means, such as passing an in-band tone on the PSTN gateway, putting
   the recording indication in a log file, or raising an application
   event in a VoiceXML dialog.  These user agents MAY also choose not to
   indicate recording awareness, thereby relying on whatever mechanism
   an SRC chooses to indicate recording, such as playing a tone in-band.

7.  SDP Handling

7.1.  Procedures at the SRC

   The SRC and SRS follows the SDP offer/answer model in [RFC3264].  The
   procedures for SRC and SRS describe the conventions used in a
   recording session.

7.1.1.  SDP handling in RS

   Since the SRC does not expect to receive media from the SRS, the SRC
   typically sets each media stream of the SDP offer to only send media,
   by qualifying them with the a=sendonly attribute, according to the
   procedures in [RFC3264].

   The SRC sends recorded streams of participants to the SRS, and the
   SRC MUST provide a label attribute (a=label), as per [RFC4574], on
   each media stream in order to identify the recorded stream with the
   rest of the metadata.  The a=label attribute identifies each recorded
   media stream, and the label name is mapped to the Media Stream
   Reference in the metadata as per [I-D.ietf-siprec-metadata].  The
   scope of the a=label attribute only applies to the SDP and Metadata
   conveyed in the bodies of the SIP request or response that the label
   appeared in.  Note that a recorded stream is distinct from a CS
   stream; the metadata provides a list of participants that contribute
   to each recorded stream.

   The following is an example SDP offer from an SRC with both audio and
   video recorded streams.  Note that the following example contains
   unfolded lines longer than 72 characters.  These are captured between
   <allOneLine> tags.





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       v=0
       o=SRC 2890844526 2890844526 IN IP4 198.51.100.1
       s=-
       c=IN IP4 198.51.100.1
       t=0 0
       m=audio 12240 RTP/AVP 0 4 8
       a=sendonly
       a=label:1
       m=video 22456 RTP/AVP 98
       a=rtpmap:98 H264/90000
       <allOneLine>
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       </allOneLine>
       a=sendonly
       a=label:2
       m=audio 12242 RTP/AVP 0 4 8
       a=sendonly
       a=label:3
       m=video 22458 RTP/AVP 98
       a=rtpmap:98 H264/90000
       <allOneLine>
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       </allOneLine>
       a=sendonly
       a=label:4



     Figure 5: Sample SDP offer from SRC with audio and video streams

7.1.1.1.  Handling media stream updates

   Over the lifetime of a recording session, the SRC can add and remove
   recorded streams from the recording session for various reasons.  For
   example, when a CS stream is added or removed from the CS, or when a
   CS is created or terminated if a recording session handles multiple
   CSes.  To remove a recorded stream from the recording session, the
   SRC sends a new SDP offer where the port of the media stream to be
   removed is set to zero, according to the procedures in [RFC3264].  To
   add a recorded stream to the recording session, the SRC sends a new
   SDP offer by adding a new media stream description or by reusing an
   old media stream which had been previously disabled, according to the
   procedures in [RFC3264].

   The SRC can temporarily discontinue streaming and collection of
   recorded media from the SRC to the SRS for reasons such as masking



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   the recording.  In this case, the SRC sends a new SDP offer and sets
   the media stream to inactive (a=inactive) for each recorded stream to
   be paused, as per the procedures in [RFC3264].  To resume streaming
   and collection of recorded media, the SRC sends a new SDP offer and
   sets the media stream to sendonly (a=sendonly).  Note that a CS
   itself may change the media stream direction by updating the SDP, for
   example, by setting a=inactive for SDP hold.  Media stream direction
   changes in CS are conveyed in the metadata by the SRC.  When a CS
   media stream is changed to/from inactive, the effect on the
   corresponding RS media stream is governed by SRC policy.  The SRC MAY
   have a local policy to pause an RS media stream when the
   corresponding CS media stream is inactive, or it MAY leave the RS
   media stream as sendonly.

7.1.2.  Recording indication in CS

   While there are existing mechanisms for providing an indication that
   a CS is being recorded, these mechanisms are usually delivered on the
   CS media streams such as playing an in-band tone or an announcement
   to the participants.  A new 'record' SDP attribute is introduced to
   allow the SRC to indicate recording state to a recording-aware UA in
   a CS.

   The 'record' SDP attribute appears at the media-level or session-
   level in either SDP offer or answer.  When the attribute is applied
   at the session-level, the indication applies to all media streams in
   the SDP.  When the attribute is applied at the media-level, the
   indication applies to the media stream only, and that overrides the
   indication if also set at the session-level.  Whenever the recording
   indication needs to change, such as termination of recording, then
   the SRC MUST initiate a reINVITE or UPDATE to update the SDP a=record
   attribute.

   The following is the ABNF of the 'record' attribute:

       attribute =/ record-attr
       ; attribute defined in RFC 4566

       record-attr = "record:" indication
       indication = "on" / "off" / "paused"

   on:  Recording is in progress.

   off:  No recording is in progress.

   paused:  Recording is in progress but media is paused.





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7.1.3.  Recording preference in CS

   When the SRC receives the a=recordpref SDP in an SDP offer or answer,
   the SRC chooses to honor the preference to record based on local
   policy at the SRC.  If the SRC makes a change in recording state, the
   SRC MUST report the new recording state in the a=record attribute in
   the SDP answer or in a subsequent SDP offer.

7.2.  Procedures at the SRS

   Typically the SRS only receives RTP streams from the SRC; therefore,
   the SDP offer/answer from the SRS normally sets each media stream to
   receive media, by setting them with the a=recvonly attribute,
   according to the procedures of [RFC3264].  When the SRS is not ready
   to receive a recorded stream, the SRS sets the media stream as
   inactive in the SDP offer or answer by setting it with an a=inactive
   attribute, according to the procedures of [RFC3264].  When the SRS is
   ready to receive recorded streams, the SRS sends a new SDP offer and
   sets the media streams with an a=recvonly attribute.

   The following is an example of an SDP answer from the SRS for the SDP
   offer from the above sample.  Note that the following example contain
   unfolded lines longer than 72 characters.  These are captured between
   <allOneLine> tags.



























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       v=0
       o=SRS 0 0 IN IP4 198.51.100.20
       s=-
       c=IN IP4 198.51.100.20
       t=0 0
       m=audio 10000 RTP/AVP 0
       a=recvonly
       a=label:1
       m=video 10002 RTP/AVP 98
       a=rtpmap:98 H264/90000
       <allOneLine>
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       </allOneLine>
       a=recvonly
       a=label:2
       m=audio 10004 RTP/AVP 0
       a=recvonly
       a=label:3
       m=video 10006 RTP/AVP 98
       a=rtpmap:98 H264/90000
       <allOneLine>
       a=fmtp:98 profile-level-id=42A01E;
                 sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
       </allOneLine>
       a=recvonly
       a=label:4



     Figure 6: Sample SDP answer from SRS with audio and video streams

   Over the lifetime of a recording session, the SRS can remove recorded
   streams from the recording session for various reasons.  To remove a
   recorded stream from the recording session, the SRS sends a new SDP
   offer where the port of the media stream to be removed is set to
   zero, according to the procedures in [RFC3264].

   The SRS MUST NOT add recorded streams in the recording session when
   the SRS sends a new SDP offer.  Similarly, when the SRS starts a
   recording session, the SRS MUST initiate the INVITE without an SDP
   offer to let the SRC generate the SDP offer with the streams to be
   recorded.

   The following sequence diagram shows an example where the SRS is
   initially not ready to receive recorded streams, and later updates
   the recording session when the SRS is ready to record.




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     SRC                                                   SRS
      |                                                     |
      |(1) INVITE (SDP offer)                               |
      |---------------------------------------------------->|
      |                                           [not ready to record]
      |                         (2)200 OK with SDP inactive |
      |<----------------------------------------------------|
      |(3) ACK                                              |
      |---------------------------------------------------->|
      |                      ...                            |
      |                                             [ready to record]
      |                     (4) re-INVITE with SDP recvonly |
      |<----------------------------------------------------|
      |(5)200 OK with SDP sendonly                          |
      |---------------------------------------------------->|
      |                                             (6) ACK |
      |<----------------------------------------------------|
      |(7) RTP                                              |
      |====================================================>|
      |                      ...                            |
      |(8) BYE                                              |
      |---------------------------------------------------->|
      |                                             (9) OK  |
      |<----------------------------------------------------|



             Figure 7: SRS responding to offer with a=inactive

7.3.  Procedures for Recording-aware User Agents

7.3.1.  Recording indication

   When a recording-aware UA receives an SDP offer or answer that
   includes the a=record attribute, the UA provides an indication to the
   end user whether the recording is on, off, or paused for each medium
   based on the most recently received a=record SDP attribute for that
   medium.

   When a CS is traversed through multiple UAs such as a B2BUA or a
   conference focus, each UA involved in the CS that is aware that the
   CS is being recorded MUST provide the recording indication through
   the a=record attribute to all other parties in the CS.

   It is possible that more than one SRC is in the call path of the same
   CS, but the recording indication attribute does not provide any hint
   as to which SRC or how many SRCs are recording.  An endpoint knows




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   only that the call is being recorded.  Furthermore, this attribute is
   not used as a request for a specific SRC to start/stop recording.

7.3.2.  Recording preference

   A participant in a CS MAY set the recording preference in the CS to
   be recorded or not recorded at session establishment or during the
   session.  A new 'recordpref' SDP attribute is introduced, and the
   participant in CS may set this recording preference attribute in any
   SDP offer/answer at session establishment time or during the session.
   The SRC is not required to honor the recording preference from a
   participant based on local policies at the SRC, and the participant
   can learn the recording indication through the a=record SDP attribute
   as described in the above section.

   The SDP a=recordpref attribute can appear at the media-level or
   session-level and can appear in an SDP offer or answer.  When the
   attribute is applied at the session-level, the recording preference
   applies to all media stream in the SDP.  When the attribute is
   applied at the media-level, the recording preference applies to the
   media stream only, and that overrides the recording preference if
   also set at the session-level.  The user agent can change the
   recording preference by changing the a=recordpref attribute in
   subsequent SDP offer or answer.  The absence of the a=recordpref
   attribute in the SDP indicates that the UA has no recording
   preference.

   The following is the ABNF of the recordpref attribute:

       attribute =/ recordpref-attr
       ; attribute defined in RFC 4566

       recordpref-attr = "a=recordpref:" pref
       pref = "on" / "off" / "pause" / "nopreference"

   on:  Sets the preference to record if it has not already been
      started.  If the recording is currently paused, the preference is
      to resume recording.

   off:  Sets the preference for no recording.  If recording has already
      been started, then the preference is to stop the recording.

   pause:  If the recording is currently in progress, sets the
      preference to pause the recording.

   nopreference:  To indicate that the UA has no preference on
      recording.




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8.  RTP Handling

   This section provides recommendations and guidelines for RTP and RTCP
   in the context of SIPREC.  In order to communicate most effectively,
   the Session Recording Client (SRC), the Session Recording Server
   (SRS), and any Recording-aware User Agents (UAs) should utilize the
   mechanisms provided by RTP in a well-defined and predicable manner.
   It is the goal of this document to make the reader aware of these
   mechanisms and provide recommendations and guidelines.

8.1.  RTP Mechanisms

   This section briefly describes important RTP/RTCP constructs and
   mechanisms that are particularly useful within the context of SIPREC.

8.1.1.  RTCP

   The RTP data transport is augmented by a control protocol (RTCP) to
   allow monitoring of the data delivery.  RTCP, as defined in
   [RFC3550], is based on the periodic transmission of control packets
   to all participants in the RTP session, using the same distribution
   mechanism as the data packets.  Support for RTCP is REQUIRED, per
   [RFC3550], and it provides, among other things, the following
   important functionality in relation to SIPREC:

   1) Feedback on the quality of the data distribution

   This feedback from the receivers may be used to diagnose faults in
   the distribution.  As such, RTCP is a well-defined and efficient
   mechanism for the SRS to inform the SRC, and for the SRC to inform
   Recording-aware UAs, of issues that arise with respect to the
   reception of media that is to be recorded.

   2) Carries a persistent transport-level identifier for an RTP source
   called the canonical name or CNAME

   The SSRC identifier may change if a conflict is discovered or a
   program is restarted, in which case receivers can use the CNAME to
   keep track of each participant.  Receivers may also use the CNAME to
   associate multiple data streams from a given participant in a set of
   related RTP sessions, for example to synchronize audio and video.
   Synchronization of media streams is also facilitated by the NTP and
   RTP timestamps included in RTCP packets by data senders.








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8.1.2.  RTP Profile

   The RECOMMENDED RTP profiles for the SRC, SRS, and Recording-aware
   UAs are "Extended Secure RTP Profile for Real-time Transport Control
   Protocol (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124], when using
   encrypted RTP streams, and "Extended RTP Profile for Real-time
   Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)"
   [RFC4585], when using non-encrypted media streams.  However, as these
   are not requirements, some implementations may use "The Secure Real-
   time Transport Protocol (SRTP)" [RFC3711], and "RTP Profile for Audio
   and Video Conferences with Minimal Control" [RFC3551].  Therefore, it
   is RECOMMENDED that the SRC, SRS, and Recording-aware UAs not rely
   entirely on RTP/SAVPF or RTP/AVPF for core functionality that may be
   at least partially achievable using RTP/SAVP and RTP/AVP.

   AVPF and SAVPF provide an improved RTCP timer model that allows more
   flexible transmission of RTCP packets in response to events, rather
   than strictly according to bandwidth.  AVPF-based codec control
   messages provide efficient mechanisms for an SRC, SRS, and Recording-
   aware UAs to handle events such as scene changes, error recovery, and
   dynamic bandwidth adjustments.  These messages are discussed in more
   detail later in this document.

   SAVP and SAVPF provide media encryption, integrity protection, replay
   protection, and a limited form of source authentication.  They do not
   contain or require a specific keying mechanism.

8.1.3.  SSRC

   The synchronization source (SSRC), as defined in [RFC3550], is
   carried in the RTP header and in various fields of RTCP packets.  It
   is a random 32-bit number that is required to be globally unique
   within an RTP session.  It is crucial that the number be chosen with
   care in order that participants on the same network or starting at
   the same time are not likely to choose the same number.  Guidelines
   regarding SSRC value selection and conflict resolution are provided
   in [RFC3550].

   The SSRC may also be used to separate different sources of media
   within a single RTP session.  For this reason as well as for conflict
   resolution, it is important that the SRC, SRS, and Recording-aware
   UAs handle changes in SSRC values and properly identify the reason of
   the change.  The CNAME values carried in RTCP facilitate this
   identification.







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8.1.4.  CSRC

   The contributing source (CSRC), as defined in [RFC3550], identifies
   the source of a stream of RTP packets that has contributed to the
   combined stream produced by an RTP mixer.  The mixer inserts a list
   of the SSRC identifiers of the sources that contributed to the
   generation of a particular packet into the RTP header of that packet.
   This list is called the CSRC list.  It is RECOMMENDED that an SRC or
   Recording-aware UA, when acting as a mixer, set the CSRC list
   accordingly, and that the SRC and SRS interpret the CSRC list per
   [RFC3550] when received.

8.1.5.  SDES

   The Source Description (SDES), as defined in [RFC3550], contains an
   SSRC/CSRC identifier followed by a list of zero or more items, which
   carry information about the SSRC/CSRC.  End systems send one SDES
   packet containing their own source identifier (the same as the SSRC
   in the fixed RTP header).  A mixer sends one SDES packet containing a
   chunk for each contributing source from which it is receiving SDES
   information, or multiple complete SDES packets if there are more than
   31 such sources.

   The ability to identify individual contributing sources is important
   in the context of SIPREC.  Metadata [I-D.ietf-siprec-metadata]
   provides a mechanism to achieve this at the signaling level.  SDES
   provides a mechanism at the RTP level.

8.1.5.1.  CNAME

   The Canonical End-Point Identifier (CNAME), as defined in [RFC3550],
   provides the binding from the SSRC identifier to an identifier for
   the source (sender or receiver) that remains constant.  It is
   important the SRC and Recording-aware UAs generate CNAMEs
   appropriately and that the SRC and SRS interpret and use them for
   this purpose.  Guidelines for generating CNAME values are provided in
   "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names
   (CNAMEs)" [RFC7022].

8.1.6.  Keepalive

   It is anticipated that media streams in SIPREC may exist in an
   inactive state for extended periods of times for any of a number of
   valid reasons.  In order for the bindings and any pinholes in NATs/
   firewalls to remain active during such intervals, it is RECOMMENDED
   that the SRC, SRS, and Recording-aware UAs follow the keep-alive
   procedure recommended in "Application Mechanism for Keeping Alive the




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   NAT Mappings Associated to RTP/RTP Control Protocol (RTCP) Flows"
   [RFC6263] for all RTP media streams.

8.1.7.  RTCP Feedback Messages

   "Codec Control Messages in the RTP Audio-Visual Profile with Feedback
   (AVPF)" [RFC5104] specifies extensions to the messages defined in
   AVPF [RFC4585].  Support for and proper usage of these messages is
   important to SRC, SRS, and Recording-aware UA implementations.  Note
   that these messages are applicable only when using the AVPF or SAVPF
   RTP profiles

8.1.7.1.  Full Intra Request

   A Full Intra Request (FIR) Command, when received by the designated
   media sender, requires that the media sender sends a Decoder Refresh
   Point at the earliest opportunity.  Using a decoder refresh point
   implies refraining from using any picture sent prior to that point as
   a reference for the encoding process of any subsequent picture sent
   in the stream.

   Decoder refresh points, especially Intra or IDR pictures for H.264
   video codecs, are in general several times larger in size than
   predicted pictures.  Thus, in scenarios in which the available bit
   rate is small, the use of a decoder refresh point implies a delay
   that is significantly longer than the typical picture duration.

8.1.7.1.1.  SIP INFO for FIR

   "XML Schema for Media Control" [RFC5168] defines an Extensible Markup
   Language (XML) Schema for video fast update.  Implementations are
   discouraged from using the method described except for backward
   compatibility purposes.  Implementations SHOULD use FIR messages
   instead.

   To make sure a common mechanism exists between the SRC and SRS, the
   SRS MUST support both mechanisms (FIR and SIP INFO), using FIR when
   negotiated successfully with the SRC, and using SIP INFO otherwise.

8.1.7.2.  Picture Loss Indicator

   Picture Loss Indication (PLI), as defined in [RFC4585], informs the
   encoder of the loss of an undefined amount of coded video data
   belonging to one or more pictures.  [RFC4585] recommends using PLI
   instead of FIR to recover from errors.  FIR is appropriate only in
   situations where not sending a decoder refresh point would render the
   video unusable for the users.  Examples where sending FIR is
   appropriate include a multipoint conference when a new user joins the



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   conference and no regular decoder refresh point interval is
   established, and a video switching MCU that changes streams.

   Appropriate use of PLI and FIR is important to ensure with minimum
   overhead that the recorded video is usable (e.g., the necessary
   reference frames exist for a player to render the recorded video).

8.1.7.3.  Temporary Maximum Media Stream Bit Rate Request

   A receiver, translator, or mixer uses the Temporary Maximum Media
   Stream Bit Rate Request (TMMBR) to request a sender to limit the
   maximum bit rate for a media stream to the provided value.
   Appropriate use of TMMBR facilitates rapid adaptation to changes in
   available bandwidth.

8.1.7.3.1.  Renegotiation of SDP bandwidth attribute

   If it is likely that the new value indicated by TMMBR will be valid
   for the remainder of the session, the TMMBR sender is expected to
   perform a renegotiation of the session upper limit using the session
   signaling protocol.  Therefore for SIPREC, implementations are
   RECOMMENDED to use TMMBR for temporary changes, and renegotiation of
   bandwidth via SDP offer/answer for more permanent changes.

8.1.8.  Symmetric RTP/RTCP for Sending and Receiving

   Within an SDP offer/answer exchange, RTP entities choose the RTP and
   RTCP transport addresses (i.e., IP addresses and port numbers) on
   which to receive packets.  When sending packets, the RTP entities may
   use the same source port or a different source port as those signaled
   for receiving packets.  When the transport address used to send and
   receive RTP is the same, it is termed "symmetric RTP" [RFC4961].
   Likewise, when the transport address used to send and receive RTCP is
   the same, it is termed "symmetric RTCP" [RFC4961].

   When sending RTP, it is REQUIRED to use symmetric RTP.  When sending
   RTCP, it is REQUIRED to use symmetric RTCP.  Although an SRS will not
   normally send RTP, it will send RTCP as well as receive RTP and RTCP.
   Likewise, although an SRC will not normally receive RTP from the SRS,
   it will receive RTCP as well as send RTP and RTCP.

      Note: Symmetric RTP and symmetric RTCP are different from RTP/RTCP
      multiplexing [RFC5761].








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8.2.  Roles

   An SRC has the task of gathering media from the various UAs in one or
   more Communication Sessions (CSs) and forwarding the information to
   the SRS within the context of a corresponding Recording Session (RS).
   There are numerous ways in which an SRC may do this, including but
   not limited to appearing as a UA within a CS, or as a B2BUA between
   UAs within a CS.

                    (Recording Session)   +---------+
                  +------------SIP------->|         |
                  |  +------RTP/RTCP----->|   SRS   |
                  |  |    +-- Metadata -->|         |
                  |  |    |               +---------+
                  v  v    |
                 +---------+
                 |   SRC   |
                 |---------| (Communication Session) +---------+
                 |         |<----------SIP---------->|         |
                 |  UA-A   |                         |  UA-B   |
                 |         |<-------RTP/RTCP-------->|         |
                 +---------+                         +---------+

                            Figure 8: UA as SRC

                                   (Recording Session)   +---------+
                                 +------------SIP------->|         |
                                 |  +------RTP/RTCP----->|   SRS   |
                                 |  |    +-- Metadata -->|         |
                                 |  |    |               +---------+
                                 v  v    |
                                +---------+
                                |   SRC   |
       +---------+              |---------|              +---------+
       |         |<----SIP----->|         |<----SIP----->|         |
       |  UA-A   |              |  B2BUA  |              |  UA-B   |
       |         |<--RTP/RTCP-->|         |<--RTP/RTCP-->|         |
       +---------+              +---------+              +---------+
             |_______________________________________________|
                          (Communication Session)

                          Figure 9: B2BUA as SRC

   The following subsections define a set of roles an SRC may choose to
   play based on its position with respect to a UA within a CS, and an
   SRS within an RS.  A CS and a corresponding RS are independent
   sessions; therefore, an SRC may play a different role within a CS
   than it does within the corresponding RS.



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8.2.1.  SRC acting as an RTP Translator

   The SRC may act as a translator, as defined in [RFC3550].  A defining
   characteristic of a translator is that it forwards RTP packets with
   their SSRC identifier intact.  There are two types of translators,
   one that simply forwards, and another that performs transcoding
   (e.g., from one codec to another) in addition to forwarding.

8.2.1.1.  Forwarding Translator

   When acting as a forwarding translator, RTP received as separate
   streams from different sources (e.g., from different UAs with
   different SSRCs) cannot be mixed by the SRC and MUST be sent
   separately to the SRS.  All RTCP reports MUST be passed by the SRC
   between the UAs and the SRS, such that the UAs and SRS are able to
   detect any SSRC collisions.

   RTCP Sender Reports generated by a UA sending a stream MUST be
   forwarded to the SRS.  RTCP Receiver Reports generated by the SRS
   MUST be forwarded to the relevant UA.

   UAs may receive multiple sets of RTCP Receiver Reports, one or more
   from other UAs participating in the CS, and one from the SRS
   participating in the RS.  A UA SHOULD process the RTCP Receiver
   Reports from the SRS if it is recording-aware.

   If SRTP is used on both the CS and the RS, decryption and/or re-
   encryption may occur.  For example, if different keys are used, it
   will occur.  If the same keys are used, it need not occur.
   Section 12 provides additional information on SRTP and keying
   mechanisms.

   If packet loss occurs, either from the UA to the SRC or from the SRC
   to the SRS, the SRS SHOULD detect and attempt to recover from the
   loss.  The SRC does not play a role in this other than forwarding the
   associated RTP and RTCP packets.

8.2.1.2.  Transcoding Translator

   When acting as a transcoding translator, an SRC MAY perform
   transcoding (e.g., from one codec to another), and this may result in
   a different rate of packets between what the SRC receives on the CS
   and what the SRC sends on the RS.  As when acting as a forwarding
   translator, RTP received as separate streams from different sources
   (e.g., from different UAs with different SSRCs) cannot be mixed by
   the SRC and MUST be sent separately to the SRS.  All RTCP reports
   MUST be passed by the SRC between the UAs and the SRS, such that the
   UAs and SRS are able to detect any SSRC collisions.



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   RTCP Sender Reports generated by a UA sending a stream MUST be
   forwarded to the SRS.  RTCP Receiver Reports generated by the SRS
   MUST be forwarded to the relevant UA.  The SRC may need to manipulate
   the RTCP Receiver Reports to take account of any transcoding that has
   taken place.

   UAs may receive multiple sets of RTCP Receiver Reports, one or more
   from other UAs participating in the CS, and one from the SRS
   participating in the RS.  A Recording-aware UA SHOULD be prepared to
   process the RTCP Receiver Reports from the SRS, whereas a recording
   unaware UA may discard such RTCP packets as not of relevance.

   If SRTP is used on both the CS and the RS, decryption and/or re-
   encryption may occur.  For example, if different keys are used, it
   will occur.  If the same keys are used, it need not occur.
   Section 12 provides additional information on SRTP and keying
   mechanisms.

   If packet loss occurs, either from the UA to the SRC or from the SRC
   to the SRS, the SRS SHOULD detect and attempt to recover from the
   loss.  The SRC does not play a role in this other than forwarding the
   associated RTP and RTCP packets.

8.2.2.  SRC acting as an RTP Mixer

   In the case of the SRC acting as a RTP mixer, as defined in
   [RFC3550], the SRC combines RTP streams from different UAs and sends
   them towards the SRS using its own SSRC.  The SSRCs from the
   contributing UA SHOULD be conveyed as CSRCs identifiers within this
   stream.  The SRC may make timing adjustments among the received
   streams and generate its own timing on the stream sent to the SRS.
   Optionally an SRC acting as a mixer can perform transcoding, and can
   even cope with different codings received from different UAs.  RTCP
   Sender Reports and Receiver Reports are not forwarded by an SRC
   acting as mixer, but there are requirements for forwarding RTCP
   Source Description (SDES) packets.  The SRC generates its own RTCP
   Sender and Receiver reports toward the associated UAs and SRS.

   The use of SRTP between the SRC and the SRS for the RS is independent
   of the use of SRTP between the UAs and SRC for the CS.  Section 12
   provides additional information on SRTP and keying mechanisms.

   If packet loss occurs from the UA to the SRC, the SRC SHOULD detect
   and attempt to recover from the loss.  If packet loss occurs from the
   SRC to the SRS, the SRS SHOULD detect and attempt to recover from the
   loss.





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8.2.3.  SRC acting as an RTP Endpoint

   The case of the SRC acting as an RTP endpoint, as defined in
   [RFC3550], is similar to the mixer case, except that the RTP session
   between the SRC and the SRS is considered completely independent from
   the RTP session that is part of the CS.  The SRC can, but need not,
   mix RTP streams from different participants prior to sending to the
   SRS.  RTCP between the SRC and the SRS is completely independent of
   RTCP on the CS.

   The use of SRTP between the SRC and the SRS for the RS is independent
   of the use of SRTP between the UAs and SRC for the CS.  Section 12
   provides additional information on SRTP and keying mechanisms.

   If packet loss occurs from the UA to the SRC, the SRC SHOULD detect
   and attempt to recover from the loss.  If packet loss occurs from the
   SRC to the SRS, the SRS SHOULD detect and attempt to recover from the
   loss.

8.3.  RTP Session Usage by SRC

   There are multiple ways that an SRC may choose to deliver recorded
   media to an SRS.  In some cases, it may use a single RTP session for
   all media within the RS, whereas in others it may use multiple RTP
   sessions.  The following subsections provide examples of basic RTP
   session usage by the SRC, including a discussion of how the RTP
   constructs and mechanisms covered previously are used.  An SRC may
   choose to use one or more of the RTP session usages within a single
   RS.  For the purpose of base interoperability between SRC and SRS, an
   SRC MUST support separate m-lines in SDP, one per CS media direction.
   The set of RTP session usages described is not meant to be
   exhaustive.

8.3.1.  SRC Using Multiple m-lines

   When using multiple m-lines, an SRC includes each m-line in an SDP
   offer to the SRS.  The SDP answer from the SRS MUST include all
   m-lines, with any rejected m-lines indicated with a zero port, per
   [RFC3264].  Having received the answer, the SRC starts sending media
   to the SRS as indicated in the answer.  Alternatively, if the SRC
   deems the level of support indicated in the answer to be
   unacceptable, it may initiate another SDP offer/answer exchange in
   which an alternative RTP session usage is negotiated.

   In order to preserve the mapping of media to participant within the
   CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to
   a unique CNAME within the RS.  Additionally, the SRC SHOULD map each
   unique combination of CNAME/SSRC within the CSs to a unique CNAME/



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   SSRC within the RS.  In doing so, the SRC may act as an RTP
   translator or as an RTP endpoint.

   The following figure illustrates a case in which each UA represents a
   participant contributing two RTP sessions (e.g., one for audio and
   one for video), each with a single SSRC.  The SRC acts as an RTP
   translator and delivers the media to the SRS using four RTP sessions,
   each with a single SSRC.  The CNAME and SSRC values used by the UAs
   within their media streams are preserved in the media streams from
   the SRC to the SRS.

                                                        +---------+
                                +------------SSRC Aa--->|         |
                                |  + --------SSRC Av--->|         |
                                |  |  +------SSRC Ba--->|   SRS   |
                                |  |  |  +---SSRC Bv--->|         |
                                |  |  |  |              +---------+
                                |  |  |  |
                                |  |  |  |
       +---------+             +----------+             +---------+
       |         |---SSRC Aa-->|   SRC    |<--SSRC Ba---|         |
       |  UA-A   |             |(CNAME-A, |             |  UA-B   |
       |(CNAME-A)|---SSRC Av-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)|
       +---------+             +----------+             +---------+

                   Figure 10: SRC Using Multiple m-lines

8.3.2.  SRC Using Mixing

   When using mixing, the SRC combines RTP streams from different
   participants and sends them towards the SRS using its own SSRC.  The
   SSRCs from the contributing participants SHOULD be conveyed as CSRCs
   identifiers.  The SRC includes one m-line for each RTP session in an
   SDP offer to the SRS.  The SDP answer from the SRS MUST include all
   m-lines, with any rejected m-lines indicated with the zero port, per
   [RFC3264].  Having received the answer, the SRC starts sending media
   to the SRS as indicated in the answer.

   In order to preserve the mapping of media to participant within the
   CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to
   a unique CNAME within the RS.  Additionally, the SRC SHOULD map each
   unique combination of CNAME/SSRC within the CSs to a unique CNAME/
   SSRC within the RS.  The SRC MUST avoid SSRC collisions, rewriting
   SSRCs if necessary when used as CSRCs in the RS.  In doing so, the
   SRC acts as an RTP mixer.






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   In the event the SRS does not support this usage of CSRC values, it
   relies entirely on the SIPREC metadata to determine the participants
   included within each mixed stream.

   The following figure illustrates a case in which each UA represents a
   participant contributing two RTP sessions (e.g., one for audio and
   one for video), each with a single SSRC.  The SRC acts as an RTP
   mixer and delivers the media to the SRS using two RTP sessions,
   mixing media from each participant into a single RTP session
   containing a single SSRC and two CSRCs.

                                          SSRC Sa       +---------+
                                  +-------CSRC Aa,Ba--->|         |
                                  |                     |         |
                                  |       SSRC Sv       |   SRS   |
                                  |   +---CSRC Av,Bv--->|         |
                                  |   |                 +---------+
                                  |   |
                               +----------+
       +---------+             |   SRC    |             +---------+
       |         |---SSRC Aa-->|(CNAME-S, |<--SSRC Ba---|         |
       |  UA-A   |             | CNAME-A, |             |  UA-B   |
       |(CNAME-A)|---SSRC Av-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)|
       +---------+             +----------+             +---------+

                        Figure 11: SRC Using Mixing

8.4.  RTP Session Usage by SRS

   An SRS that supports recording an audio CS MUST support SRC usage of
   separate audio m-lines in SDP, one per CS media direction.  An SRS
   that supports recording a video CS MUST support SRC usage of separate
   video m-lines in SDP, one per CS media direction.  Therefore, for an
   SRS supporting a typical audio call, the SRS has to support receiving
   at least two audio m-lines.  For an SRS supporting a typical audio
   and video call, the SRS has to support receiving at least four total
   m-lines in the SDP, two audio m-lines and two video m-lines.

   These requirements allow an SRS to be implemented that supports video
   only, without requiring support for audio recording.  They also allow
   an SRS to be implemented that supports recording only one direction
   of one stream in a CS; for example, an SRS designed to record
   security monitoring cameras that only send (not receive) video
   without any audio.  These requirements were not written to prevent
   other modes being implemented and used, such as using a single m-line
   and mixing the separate audio streams together.  Rather, the
   requirements were written to provide a common base mode to implement
   for the sake of interoperability.  It is important to note that an



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   SRS implementation supporting the common base may not record all
   media streams in a CS if a participant supports more than one m-line
   in a video call, such as one for camera and one for presentation.
   SRS implementations may support other modes as well, but have to at
   least support the ones above such that they interoperate in the
   common base mode for basic interoperability.

9.  Metadata

   Some metadata attributes are contained in SDP, and others are
   contained in a new content type "application/rs-metadata".  The
   format of the metadata is described as part of the mechanism in
   [I-D.ietf-siprec-metadata].  A new "disposition-type" of Content-
   Disposition is defined for the purpose of carrying metadata.  The
   value is "recording-session", which indicates the "application/rs-
   metadata" content contains metadata to be handled by the SRS.

9.1.  Procedures at the SRC

   The SRC MUST send metadata to the SRS in an RS.  The SRC SHOULD send
   metadata as soon as it becomes available and whenever it changes.
   Cases in which an SRC may be justified in waiting temporarily before
   sending metadata include:

   o  waiting for a previous metadata exchange to complete (i.e., the
      SRC cannot send another SDP offer until the previous offer/answer
      completes, and may prefer not to send an UPDATE during this time
      either).

   o  constraining the signaling rate on the RS.

   o  sending metadata when key events occur rather than for every event
      that has any impact on metadata.

   The SRC may also be configured to suppress certain metadata out of
   concern for privacy or perceived lack of need for it to be included
   in the recording.

   Metadata sent by the SRC is categorized as either a full metadata
   snapshot or a partial update.  A full metadata snapshot describes all
   metadata associated with the RS.  The SRC MAY send a full metadata
   snapshot at any time.  The SRC MAY send a partial update only if a
   full metadata snapshot has been sent previously.

   The SRC MAY send metadata (either a full metadata snapshot or a
   partial update) in an INVITE request, an UPDATE request [RFC3311], or
   a 200 response to an offerless INVITE from the SRS.  If the metadata




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   contains a reference to any SDP labels, the request containing the
   metadata MUST also contain an SDP offer that defines those labels.

   When a SIP message contains both an SDP offer and metadata, the
   request body MUST have content type "multipart/mixed", with one
   subordinate body part containing the SDP offer and another containing
   the metadata.  When a SIP message contains only an SDP offer or
   metadata, the "multipart/mixed" container is optional.

   The SRC SHOULD include a full metadata snapshot in the initial INVITE
   request establishing the RS.  If metadata is not yet available (e.g.,
   an RS established in absence of a CS), the SRC SHOULD send a full
   metadata snapshot as soon as metadata becomes available.

   If the SRC receives a snapshot request from the SRS, it MUST
   immediately send a full metadata snapshot.

   The following is an example of a full metadata snapshot sent by the
   SRC in the initial INVITE request:
































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       INVITE sip:recorder@example.com SIP/2.0
       Via: SIP/2.0/TCP src.example.com;branch=z9hG4bKdf6b622b648d9
       From: <sip:2000@example.com>;tag=35e195d2-947d-4585-946f-09839247
       To: <sip:recorder@example.com>
       Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
       CSeq: 101 INVITE
       Max-Forwards: 70
       Require: siprec
       Accept: application/sdp, application/rs-metadata
       Contact: <sip:2000@src.example.com>;+sip.src
       Content-Type: multipart/mixed;boundary=foobar
       Content-Length: [length]

       --foobar
       Content-Type: application/sdp

       v=0
       o=SRS 2890844526 2890844526 IN IP4 198.51.100.1
       s=-
       c=IN IP4 198.51.100.1
       t=0 0
       m=audio 12240 RTP/AVP 0 4 8
       a=sendonly
       a=label:1

       --foobar
       Content-Type: application/rs-metadata
       Content-Disposition: recording-session

       [metadata content]



        Figure 12: Sample INVITE request for the recording session

9.2.  Procedures at the SRS

   The SRS receives metadata updates from the SRC in INVITE and UPDATE
   requests.  Since the SRC can send partial updates based on the
   previous update, the SRS needs to keep track of the sequence of
   updates from the SRC.

   In the case of an internal failure at the SRS, the SRS may fail to
   recognize a partial update from the SRC.  The SRS may be able to
   recover from the internal failure by requesting a full metadata
   snapshot from the SRC.  Certain errors, such as syntax errors or
   semantic errors in the metadata information, are likely caused by an
   error on the SRC side, and it is likely the same error will occur



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   again even when a full metadata snapshot is requested.  In order to
   avoid repeating the same error, the SRS can simply terminate the
   recording session when a syntax error or semantic error is detected
   in the metadata.

   The SRS MAY explicitly request a full metadata snapshot by sending an
   UPDATE request.  This request MUST contain a body with content
   disposition type "recording-session", and MUST NOT contain an SDP
   body.  The SRS MUST NOT request a full metadata snapshot in an UPDATE
   response or in any other SIP transaction.  The format of the content
   is "application/rs-metadata", and the body is an XML document, the
   format of which is defined in [I-D.ietf-siprec-metadata].  The
   following shows an example:


     UPDATE sip:2000@src.exmaple.com SIP/2.0
     Via: SIP/2.0/UDP srs.example.com;branch=z9hG4bKdf6b622b648d9
     To: <sip:2000@exmaple.com>;tag=35e195d2-947d-4585-946f-098392474
     From: <sip:recorder@example.com>;tag=1234567890
     Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
     CSeq: 1 UPDATE
     Max-Forwards: 70
     Require: siprec
     Contact: <sip:recorder@srs.example.com>;+sip.srs
     Accept: application/sdp, application/rs-metadata
     Content-Disposition: recording-session
     Content-Type: application/rs-metadata
     Content-Length: [length]

     <?xml version="1.0" encoding="UTF-8"?>
       <requestsnapshot xmlns='urn:ietf:params:xml:ns:recording:1'>
         <requestreason xml:lang="it">SRS internal error</requestreason>
       </requestsnapshot>



                        Figure 13: Metadata Request

   Note that UPDATE was chosen for the SRS to request metadata snapshot
   because it can be sent regardless of the state of the dialog.  This
   was seen as better than requiring support for both UPDATE and re-
   INVITE for this operation.

   When the SRC receives a request for a metadata snapshot, it MUST
   immediately provide a full metadata snapshot in a separate INVITE or
   UPDATE transaction.  Any subsequent partial updates will not be
   dependent on any metadata sent prior to this full metadata snapshot.




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   The metadata received by the SRS can contain ID elements used to
   cross reference one element to another.  An element containing the
   definition of an ID, and an element containing a reference to that ID
   will often be received from the same SRC.  It is also valid for those
   elements to be received from different SRCs, for example, when each
   endpoint in the same CS act as an SRC to record the call and a common
   ID refers to the same CS.  The SRS MUST NOT consider this an error.

10.  Persistent Recording

   Persistent recording is a specific use case outlined in REQ-005 or
   Use Case 4 in [RFC6341], where a recording session can be established
   in the absence of a communication session.  The SRC continuously
   records media in a recording session to the SRS even in the absence
   of a CS for all user agents that are part of persistent recording.
   By allocating recorded streams and continuously sending recorded
   media to the SRS, the SRC does not have to prepare new recorded
   streams with a new SDP offer when a new communication session is
   created and also does not impact the timing of the CS.  The SRC only
   needs to update the metadata when new communication sessions are
   created.

   When there is no communication session running on the devices with
   persistent recording, there is no recorded media to stream from the
   SRC to the SRS.  In certain environments where Network Address
   Translator (NAT) is used, typically a minimum of flow activity is
   required to maintain the NAT binding for each port opened.  Agents
   that support Interactive Connectivity Establishment (ICE) solve this
   problem.  For non-ICE agents, in order not to lose the NAT bindings
   for the RTP/RTCP ports opened for the recorded streams, the SRC and
   SRS SHOULD follow the recommendations provided in [RFC6263] to
   maintain the NAT bindings.

11.  IANA Considerations

11.1.  Registration of Option Tags

   This specification registers two option tags.  The required
   information for this registration, as specified in [RFC3261], is as
   follows.

11.1.1.  siprec Option Tag

      Name: siprec

      Description: This option tag is for identifying that the SIP
      session is for the purpose of a recording session.  This is
      typically not used in a Supported header.  When present in a



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      Require header in a request, it indicates that the UA is either an
      SRC or SRS capable of handling a recording session.

11.1.2.  record-aware Option Tag

      Name: record-aware

      Description: This option tag is to indicate the ability for the
      user agent to receive recording indicators in media-level or
      session-level SDP.  When present in a Supported header, it
      indicates that the UA can receive recording indicators in media-
      level or session-level SDP.

11.2.  Registration of media feature tags

   This document registers two new media feature tags in the SIP tree
   per the process defined in [RFC2506] and [RFC3840]

11.2.1.  src feature tag

      Media feature tag name: sip.src

      ASN.1 Identifier: TBD at registration

      Summary of the media feature indicated by this tag: This feature
      tag indicates that the user agent is a Session Recording Client
      for the purpose of a Recording Session.

      Values appropriate for use with this feature tag: boolean

      The feature tag is intended primarily for use in the following
      applications, protocols, services, or negotiation mechanisms: This
      feature tag is only useful for a Recording Session.

      Examples of typical use: Routing the request to a Session
      Recording Server.

      Security Considerations: Security considerations for this media
      feature tag are discussed in Section 11.1 of RFC 3840.

11.2.2.  srs feature tag

      Media feature tag name: sip.srs

      ASN.1 Identifier: TBD at registration






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      Summary of the media feature indicated by this tag: This feature
      tag indicates that the user agent is a Session Recording Server
      for the purpose of a Recording Session.

      Values appropriate for use with this feature tag: boolean

      The feature tag is intended primarily for use in the following
      applications, protocols, services, or negotiation mechanisms: This
      feature tag is only useful for a Recording Session.

      Examples of typical use: Routing the request to a Session
      Recording Client.

      Security Considerations: Security considerations for this media
      feature tag are discussed in Section 11.1 of RFC 3840.

11.3.  New Content-Disposition Parameter Registrations

   This document registers a new "disposition-type" value in Content-
   Disposition header: recording-session.

   recording-session:  The body describes either:

      *  metadata about the recording session

      *  reason for metadata snapshot request

      as determined by the MIME value indicated in the Content-Type.

11.4.  Media Type Registration

11.5.  SDP Attributes

   This document registers the following new SDP attributes.

11.5.1.  'record' SDP Attribute

   Contact names: Leon Portman leon.portman@gmail.com, Henry Lum
   henry.lum@genesyslab.com

   Attribute name: record

   Long form attribute name: Recording Indication

   Type of attribute: session or media-level

   Subject to charset: no




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   This attribute provides the recording indication for the session or
   media stream.

   Allowed attribute values: on, off, paused

11.5.2.  'recordpref' SDP Attribute

   Contact names: Leon Portman leon.portman@nice.com, Henry Lum
   henry.lum@genesyslab.com

   Attribute name: recordpref

   Long form attribute name: Recording Preference

   Type of attribute: session or media-level

   Subject to charset: no

   This attribute provides the recording preference for the session or
   media stream.

   Allowed attribute values: on, off, pause, nopreference

12.  Security Considerations

   The recording session is fundamentally a standard SIP dialog
   [RFC3261]; therefore, the recording session can reuse any of the
   existing SIP security mechanisms available for securing the session
   signaling, the recorded media, and the metadata.  The use cases and
   requirements document [RFC6341] outlines the general security
   considerations, and this document describes specific security
   recommendations.

   The SRC and SRS MUST support SIP with TLS version 1.2, SHOULD follow
   the best practices when using TLS as per [RFC7525], and MAY use SIPS
   with TLS as per [RFC5630].  The Recording Session MUST be at least as
   secure as the Communication Session, meaning using at least the same
   strength of cipher suite as the CS if the CS is secured.  For
   example, if the CS uses SIPS for signaling and RTP/SAVP for media,
   then the RS may not use SIP or plain RTP unless other equivalent
   security measures are in effect, since doing so would mean an
   effective security downgrade.  Examples of other potentially
   equivalent security mechanisms include mutually-authenticated TLS for
   the RS signaling channel or an appropriately protected network path
   for the RS media component.






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12.1.  Authentication and Authorization

   At the transport level, the recording session uses TLS authentication
   to validate the authenticity of the SRC and SRS.  The SRC and SRS
   MUST implement TLS mutual authentication for establishing the
   recording session.  Whether the SRC/SRS chooses to use TLS mutual
   authentication is a deployment decision.  In deployments where a UA
   acts as its own SRC, this requires the UA have its own certificate as
   needed for TLS mutual authentication.  In deployments where the SRC
   and the SRS are in the same administrative domain and have some other
   means of assuring authenticity, the SRC and SRS may choose not to
   authenticate each other, or to have the SRC authenticate the SRS
   only.  In deployments where the SRS can be hosted on a different
   administrative domain, it is important to perform mutual
   authentication to ensure the authenticity of both the SRC and the SRS
   before transmitting any recorded media.  The risk of not
   authenticating the SRS is that the recording may be sent to an entity
   other than the intended SRS, allowing a sensitive call recording to
   be received by an attacker.  On the other hand, the risk of not
   authenticating the SRC is that an SRS will accept calls from an
   unknown SRC and allow potential forgery of call recordings.

   There may be scenarios in which the signaling between the SRC and SRS
   is not direct, e.g., a SIP proxy exists between the SRC and the SRS.
   In such scenarios, each hop is subject to the TLS mutual
   authentication constraint and transitive trust at each hop is
   utilized.  Additionally, an SRC or SRS may use other existing SIP
   mechanisms available, including but not limited to, Digest
   Authentication [RFC3261], Asserted Identity [RFC3325], and Connected
   Identity [RFC4916].

   The SRS may have its own set of recording policies to authorize
   recording requests from the SRC.  The use of recording policies is
   outside the scope of the Session Recording Protocol.

12.2.  RTP handling

   In many scenarios it will be critical for the media transported
   between the SRC and the SRS to be protected.  Media encryption is an
   important element in the overall SIPREC solution; therefore the SRC
   and the SRS MUST support RTP/SAVP [RFC3711] and RTP/SAVPF [RFC5124].
   RTP/SAVP and RTP/SAVPF provide media encryption, integrity
   protection, replay protection, and a limited form of source
   authentication.  They do not contain or require a specific keying
   mechanism.  At a minimum, the SRC and SRS MUST support the SDP
   Security Descriptions (SDES) key negotiation mechanism [RFC4568].
   For cases in which DTLS-SRTP is used to encrypt a CS media stream, an
   SRC may use SRTP Encrypted Key Transport (EKT)



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   [I-D.ietf-avtcore-srtp-ekt] in order to use SRTP-SDES in the RS
   without needing to re-encrypt the media.

      Note: When using EKT in this manner, it is possible for
      participants in the CS to send traffic that appears to be from
      other participants and have this forwarded by the SRC to the SRS
      within the RS.  If this is a concern (e.g. the RS is intended for
      audit or compliance purposes), EKT is not an appropriate choice.

   When RTP/SAVP or RTP/SAVPF is used, an SRC can choose to use the same
   or different keys in the RS than the ones used in the CS.  Some SRCs
   are designed to simply replicate RTP packets from a CS media stream
   to the SRS, in which case the SRC will use the same key in the RS as
   used in the CS.  In this case, the SRC MUST secure the SDP containing
   the keying material in the RS with at least the same level of
   security as in the CS.  The risk of lowering the level of security in
   the RS is that it will effectively become a downgrade attack on the
   CS since the same key is used for both CS and RS.

   SRCs that decrypt an encrypted CS media stream and re-encrypt it when
   sending it to the SRS MUST use a different key than what is used for
   the CS media stream, to ensure that it is not possible for someone
   who has the key for the CS media stream to access recorded data they
   are not authorized to access.  In order to maintain a comparable
   level of security, the key used in the RS SHOULD of equivalent or
   greater strength than that used in the CS.

12.3.  Metadata

   Metadata contains sensitive information such as the address of record
   of the participants and other extension data placed by the SRC.  It
   is essential to protect the content of the metadata in the RS.  Since
   metadata is a content type transmitted in SIP signaling, metadata
   SHOULD be protected at the transport level by SIPS/TLS.

12.4.  Storage and playback

   While storage and playback of the call recording is beyond the scope
   of this document, it is worthwhile to mention here that it is also
   important for the recording storage and playback to provide a level
   of security that is comparable to the communication session.  It
   would defeat the purpose of securing both the communication session
   and the recording session mentioned in the previous sections if the
   recording can be easily played back with a simple, unsecured HTTP
   interface without any form of authentication or authorization.






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13.  Acknowledgements

   We want to thank John Elwell, Paul Kyzivat, Partharsarathi R, Ram
   Mohan R, Hadriel Kaplan, Adam Roach, Miguel Garcia, Thomas Stach,
   Muthu Perumal, Dan Wing, and Magnus Westerlund for their valuable
   comments and inputs to this document.

14.  References

14.1.  Normative References

   [I-D.ietf-siprec-metadata]
              R, R., Ravindran, P., and P. Kyzivat, "Session Initiation
              Protocol (SIP) Recording Metadata", draft-ietf-siprec-
              metadata-18 (work in progress), August 2015.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <http://www.rfc-editor.org/info/rfc2119>.

   [RFC2506]  Holtman, K., Mutz, A., and T. Hardie, "Media Feature Tag
              Registration Procedure", BCP 31, RFC 2506,
              DOI 10.17487/RFC2506, March 1999,
              <http://www.rfc-editor.org/info/rfc2506>.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              DOI 10.17487/RFC3261, June 2002,
              <http://www.rfc-editor.org/info/rfc3261>.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,
              <http://www.rfc-editor.org/info/rfc3264>.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <http://www.rfc-editor.org/info/rfc3550>.

   [RFC3840]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
              "Indicating User Agent Capabilities in the Session
              Initiation Protocol (SIP)", RFC 3840,
              DOI 10.17487/RFC3840, August 2004,
              <http://www.rfc-editor.org/info/rfc3840>.




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   [RFC4574]  Levin, O. and G. Camarillo, "The Session Description
              Protocol (SDP) Label Attribute", RFC 4574,
              DOI 10.17487/RFC4574, August 2006,
              <http://www.rfc-editor.org/info/rfc4574>.

   [RFC7245]  Hutton, A., Ed., Portman, L., Ed., Jain, R., and K. Rehor,
              "An Architecture for Media Recording Using the Session
              Initiation Protocol", RFC 7245, DOI 10.17487/RFC7245, May
              2014, <http://www.rfc-editor.org/info/rfc7245>.

14.2.  Informative References

   [I-D.ietf-avtcore-srtp-ekt]
              Mattsson, J., McGrew, D., and D. Wing, "Encrypted Key
              Transport for Secure RTP", draft-ietf-avtcore-srtp-ekt-03
              (work in progress), October 2014.

   [RFC2804]  IAB and , "IETF Policy on Wiretapping", RFC 2804,
              DOI 10.17487/RFC2804, May 2000,
              <http://www.rfc-editor.org/info/rfc2804>.

   [RFC3311]  Rosenberg, J., "The Session Initiation Protocol (SIP)
              UPDATE Method", RFC 3311, DOI 10.17487/RFC3311, October
              2002, <http://www.rfc-editor.org/info/rfc3311>.

   [RFC3325]  Jennings, C., Peterson, J., and M. Watson, "Private
              Extensions to the Session Initiation Protocol (SIP) for
              Asserted Identity within Trusted Networks", RFC 3325,
              DOI 10.17487/RFC3325, November 2002,
              <http://www.rfc-editor.org/info/rfc3325>.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              DOI 10.17487/RFC3551, July 2003,
              <http://www.rfc-editor.org/info/rfc3551>.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, DOI 10.17487/RFC3711, March 2004,
              <http://www.rfc-editor.org/info/rfc3711>.

   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
              <http://www.rfc-editor.org/info/rfc4568>.






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   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              DOI 10.17487/RFC4585, July 2006,
              <http://www.rfc-editor.org/info/rfc4585>.

   [RFC4916]  Elwell, J., "Connected Identity in the Session Initiation
              Protocol (SIP)", RFC 4916, DOI 10.17487/RFC4916, June
              2007, <http://www.rfc-editor.org/info/rfc4916>.

   [RFC4961]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
              BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,
              <http://www.rfc-editor.org/info/rfc4961>.

   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
              February 2008, <http://www.rfc-editor.org/info/rfc5104>.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
              2008, <http://www.rfc-editor.org/info/rfc5124>.

   [RFC5168]  Levin, O., Even, R., and P. Hagendorf, "XML Schema for
              Media Control", RFC 5168, DOI 10.17487/RFC5168, March
              2008, <http://www.rfc-editor.org/info/rfc5168>.

   [RFC5630]  Audet, F., "The Use of the SIPS URI Scheme in the Session
              Initiation Protocol (SIP)", RFC 5630,
              DOI 10.17487/RFC5630, October 2009,
              <http://www.rfc-editor.org/info/rfc5630>.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761,
              DOI 10.17487/RFC5761, April 2010,
              <http://www.rfc-editor.org/info/rfc5761>.

   [RFC6263]  Marjou, X. and A. Sollaud, "Application Mechanism for
              Keeping Alive the NAT Mappings Associated with RTP / RTP
              Control Protocol (RTCP) Flows", RFC 6263,
              DOI 10.17487/RFC6263, June 2011,
              <http://www.rfc-editor.org/info/rfc6263>.

   [RFC6341]  Rehor, K., Ed., Portman, L., Ed., Hutton, A., and R. Jain,
              "Use Cases and Requirements for SIP-Based Media Recording
              (SIPREC)", RFC 6341, DOI 10.17487/RFC6341, August 2011,
              <http://www.rfc-editor.org/info/rfc6341>.



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   [RFC7022]  Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
              September 2013, <http://www.rfc-editor.org/info/rfc7022>.

   [RFC7525]  Sheffer, Y., Holz, R., and P. Saint-Andre,
              "Recommendations for Secure Use of Transport Layer
              Security (TLS) and Datagram Transport Layer Security
              (DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May
              2015, <http://www.rfc-editor.org/info/rfc7525>.

Authors' Addresses

   Leon Portman
   NICE Systems
   22 Zarhin Street
   P.O. Box 690
   Ra'anana  4310602
   Israel

   Email: leon.portman@gmail.com


   Henry Lum (editor)
   Genesys
   1380 Rodick Road, Suite 201
   Markham, Ontario  L3R4G5
   Canada

   Email: henry.lum@genesyslab.com


   Charles Eckel
   Cisco
   170 West Tasman Drive
   San Jose, CA 95134
   United States

   Email: eckelcu@cisco.com


   Alan Johnston
   Avaya
   St. Louis, MO  63124

   Email: alan.b.johnston@gmail.com





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   Andrew Hutton
   Unify
   Brickhill Street
   Milton Keynes  MK15 0DJ
   United Kingdom

   Email: andrew.hutton@unify.com












































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